Merge "ASoC: pcm: QGKI startup sequence for playback and capture" (e1bdeb41) · Commits · e / devices / android_kernel_fairphone_FP5

sound/soc/soc-pcm.c

+9 −1

Original line number	Diff line number	Diff line
		@@ -801,11 +801,12 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)

		mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);

		#ifdef CONFIG_AUDIO_QGKI
		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
		snd_soc_dapm_stream_event(rtd,
		SNDRV_PCM_STREAM_PLAYBACK,
		SND_SOC_DAPM_STREAM_START);

		#endif
		if (rtd->dai_link->ops->prepare) {
		ret = rtd->dai_link->ops->prepare(substream);
		if (ret < 0) {
		@@ -850,6 +851,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
		cancel_delayed_work(&rtd->delayed_work);
		}

		#ifdef CONFIG_AUDIO_QGKI
		if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
		for (i = 0; i < rtd->num_codecs; i++) {
		codec_dai = rtd->codec_dais[i];
		@@ -859,6 +861,10 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
		SND_SOC_DAPM_STREAM_START);
		}
		}
		#else
		snd_soc_dapm_stream_event(rtd, substream->stream,
		SND_SOC_DAPM_STREAM_START);
		#endif

		for_each_rtd_codec_dai(rtd, i, codec_dai)
		snd_soc_dai_digital_mute(codec_dai, 0,
		@@ -866,6 +872,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
		snd_soc_dai_digital_mute(cpu_dai, 0, substream->stream);

		out:
		#ifdef CONFIG_AUDIO_QGKI
		if (ret < 0 && substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
		pr_err("%s: Issue stop stream for codec_dai due to op failure %d = ret\n",
		__func__, ret);
		@@ -873,6 +880,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
		SNDRV_PCM_STREAM_PLAYBACK,
		SND_SOC_DAPM_STREAM_STOP);
		}
		#endif
		mutex_unlock(&rtd->card->pcm_mutex);

		return ret;